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300-075 Implementing Cisco IP Telephony and Video, fraction 2 (CIPTV2)

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300-075 exam Dumps Source : Implementing Cisco IP Telephony and Video, fraction 2 (CIPTV2)

Test Code : 300-075
Test appellation : Implementing Cisco IP Telephony and Video, fraction 2 (CIPTV2)
Vendor appellation : Cisco
free pdf : 649 existent Questions

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Cisco Implementing Cisco IP Telephony

Cisco IP Telephony (training) | killexams.com existent Questions and Pass4sure dumps

Cisco's IP telephony offering has become so complicated in such a brief length of time that to find out about it now includes a two-part, eight-day working towards route. Delegates can both attend the two parts as a standalone option (and pick the course exam if they opt for), or pick a further four lessons and fade for the Cisco certified Voice professional (CCVP) qualification.

global competencies is an reliable practicing companion of Cisco, and so intently follows its curriculum. It is among the largest training suppliers for Cisco in the UK, having acquired the ragged biggest company, Azlan, in March 2006. because of the acquisition, it boasts a purview of working towards centres throughout the South East, the Midlands and the North.

The course specializes in CallManager (now called Cisco Unified Communications manager), the call-processing software in Cisco's IP telephony providing. Delegates can choose to study CallManager edition 4, the most widely deployed windows-based edition, or edition 5, which is Linux-based mostly.

ZDNet.co.uk reviewed the first a fraction of the eight-day course and chose to focal point on edition four. the first fraction lasts 5 days and covers about five-eighths of the content. Delegates are given dates for the 2d half, which is three days in size, presently after the conclusion of fraction one.

LocationWe reviewed this course at world potential's London training centre, a diligent basement facility in the heart of the metropolis's enterprise district. international odds besides teaches the route in its Wakefield, Coventry and Wokingham working towards centres.

PrerequisitesGlobal odds suggests that delegates maintain already studied for each the Cisco certified Networking associate (CCNA) and Cisco Voice over IP (CVoice) certifications. however, basically, a number of years of journey in both statistics and voice communications is probably going to subsist sufficient.

topics coveredCallManager working gadget, database and supporting functions; CallManager deployment alternatives; setting up a call; Cisco telephony endpoints; CallManager configuration; configuring leaven switches to wield voice; configuring Cisco IP Communicator gentle phone; the usage of the bulk administration device; configuring gateways and trunks; configuring route plans; configuring hunt groups; imposing category of handle; multi-website deployments; enabling points; configuring CallManager purposes; adding video; first-rate of provider; protection; troubleshooting.

DeliveryThis path changed into delivered by using an extremely knowledgeable tutor who understood Cisco's IP telephony gear fully. He changed into hardly ever unable to reply a question and he populated the course with factual titbits, some of that maintain been positive in functional phrases ('bear in sarcasm that passwords are reset when a CallManager update is applied') and some of that maintain been just simple wonderful ('did you know why the MAC wield carries that prefix?'). significantly, he gave close pointers to the category of questions that may loom within the examination.

despite the fact the tutor is permanently employed by world talents, he said that he spends 20 percent of his working time outside the school margin so as to sustain-to-date with IP and Cisco expertise. His title appeared to subsist substantiated, as he talked at size about trade hobbies and functional experiences.

The tutor besides ran useful workouts configuring CallManager, which accounted for approximately 60 p.c of the route's period. each and every step was clearly described in precise detail in the course notes, with the influence that dinky thought was required to follow the notes, arguably impeding the discovering manner.

We had only 1 criticism of the tutor: however he held the delegates' attention admirably, he did, during this reviewer's opinion, fail to wholly engage with the delegates in two-means communication.

best of aiding materialsDelegates were provided with three hefty units of notes overlaying the tutor's shows and useful workouts. These had been for essentially the most fraction provided through Cisco, with little local additions. this is typical observe for a technology training course. Delegates maintain been besides offered free ongoing lead with the aid of electronic mail.

cost-addOur judgement on the charge of the path for odds delegates comes with one caveat. Delegates ought to accept as just with the chance that they're going to deserve to install an IP telephony providing it really is made by Cisco. in terms of configuration, Cisco's IP telephony machine is significantly distinctive to any other seller's device, and so the path handiest has tremendous charge for Cisco deployments. IT specialists who're planning to install non-Cisco IP telephony could conclude more desirable in deciding on one of the many widely wide-spread IP-telephony classes on present from different suppliers.

however, if the delegate is import to deploy Cisco package, then the price-add from the eight days is considerable: an vital amount of hands-on potential may besides subsist taken away.

Certification and additional opportunitiesDelegates can determine to sit down an examination after the discontinue of the direction if they hope, and might then proceed to complete the CCVP if they desire. The other lessons inside the CCVP are every sole provided via international skills. they are: worthy of provider; Cisco Voice over IP; Troubleshooting Cisco Unified Communications methods; enforcing Cisco Voice Gateways and Gatekeepers.

however a notice of warning: the total fees to attend every sole of the classes faultfinding for the CCVP will set delegates again a color over £10,000 — and that is the reason devoid of examination prices or VAT. The certification can besides subsist held for 3 years before re-testing is required.

students might then choose to analyze for the much sought-after certified Cisco Internetwork skilled (CCIE) qualification, Cisco's best certification.


New Cisco Cert objectives affluent Media | killexams.com existent Questions and Pass4sure dumps

Citing stories claiming billion greenback boom in on-demand media, Cisco has announced a new certification for affluent media in IP networks.

in keeping with the company, its new certification, affluent Media Communications specialist, will subsist confident IT professionals "have the practising to forward the skills required to design, deploy and manage prosperous media functions to wield new market calls for, including integrating video conferencing functions right into a voice and records network."

Cisco items the certification encompasses will consist of Cisco CallManager for IP telephony, Cisco harmony voice messaging and Cisco IP Contact facilities.

earning a affluent Media professional certification will require conserving an lively Cisco certified network affiliate (CCNA) certification and passing two extra assessments:

  • 642-481 CRMC -- enforcing Cisco IP Video convention affluent Media Communications (ICRMC)
  • 642-432 CVOICE -- Cisco Voice Over IP (CVOICE)
  • The trade is providing the implementing Cisco IP Video conference wealthy Media Communications (ICRMC) path, a self-paced category it expects most students will finished in about eight hours. The company's informed practicing for CVOICE is attainable as a 5-day instructor-led type, and as a few e-discovering options.


    Cisco Collaboration technology applied to Agribusiness in Brazil | killexams.com existent Questions and Pass4sure dumps

    Sao Paulo, Brazil, December eight, 2015 - SLC Agricola, a Brazilian agricultural commodities producer, has adopted Cisco® collaboration technology to upgrade its conversation infrastructure. The enterprise deployed Cisco IP phones, video conferencing solutions and speedy messaging applications to connect its headquarters and 14 construction sites. benefits from the expertise adoption encompass decreased mobilephone and transport expenses besides stronger communications and superior staff productivity.

    headquartered in 1977 by pass of the SLC group, SLC Agricola changed into one of the vital first Brazilian organizations specializing in the grains and cotton sector to subsist listed on the stock market, fitting a benchmark in the phase. The enterprise operates 14 farms across six Brazilian states and is in consistent growth, so the ancient telephony platform became saturated. furthermore, lots of the production facilities can subsist found in faraway locations, so the issue in deploying terminals and keeping the analog device became expensive and counterproductive.

    The Cisco’s associate Infra TI led the telecomm integration task, and after an entire evaluation on the entire SLC Agricola environment, it designed the migration scheme to the IP system. step one become conducted at SLC's headquarters in Porto Alegre, by pass of implementing Cisco CallManager servers and more than 200 consumer extensions. Two carrier portfolios were built: a typical one with Cisco Unified SIP mobile 3905 contraptions for the team of workers; and a further one for managers and heads of departments, with Cisco Unified IP telephone 8945 capabilities, including brief-start videoconferencing.

    The Cisco Jabber® software, which allows mixed rapidly messaging (IM), voice and video, and Cisco freeway were besides carried out to aid using cell gadgets without a VPN.

    "the brand new platform has vastly contributed to decreased prices and streamlined production, and we've several examples to prove it," referred to João Aranda, telecom coordinator at SLC Agricola. "To identify one, when the SLC’s presidential group of workers became visiting farm, they were capable of attend an pressing meeting via Cisco video conferencing, saving time and fade back and forth charges, and warding off any distress to the resolution-making manner," he talked about.

    "Collaboration options maintain a different utility to agribusiness," talked about Ana Claudia Plihal, industrial director at Cisco. "IP expertise has the vigour to eliminate the exorbitant cabling of established telephony, simplifying the operation in far off environments. At SLC, for instance, the brand new gear will no longer interfere within the farm’s infrastructure. It’s a simple case of plugging new telephones to the gateway of every area," she pointed out. "additionally, the probability to maintain video conferencing available anytime not handiest adds value to the trade but besides presents the enterprise an imaginative profile."

    The SLC Agrícola assignment began in 2014, and these days every sole farms already maintain at the least one video conferencing unit apportion in. additional benefits highlighted via the assignment group encompass simplified administration and adjustments in the corporate lifestyle, which directly influenced the resolution-making technique.

    Aranda delivered that discounts made on the headquarters, production units and by the organization’s executives with decreased mobilephone and shuttle prices had been no longer calculated yet; although, the consequences completed with the brand new IP expertise are gigantic. "we maintain actually had considerable positive aspects…. for example, many of their internal procedures are now obtainable by pass of IP telephony, comparable to workforce practising and the organization’s trainee program.” SLC Agricola’s telecommunications help is anticipated to subsist accomplished inside two to three years.

    About SLC Agricola

    SLC Agricola, is an agricultural producer, focusing chiefly on cotton, soybeans and corn. The trade operates 14 farms strategically determined in six Brazilian states totaling 343.6 thousand hectares within the 2013/14 advertising 12 months, and a forecast of planted area of 370.1 thousand hectares for the 2014/15 crop, being 206.9 thousand hectares planted with soybeans, 98.6 thousand with cotton, 43.1 thousand with corn, and 18.1 thousand hectares with other crops reminiscent of espresso, wheat, corn seed and sugarcane. The enterprise’s enterprise model is in line with a big-scale, coincident creation system, with standardized production devices, leading know-how, strict cost controls, and environmental accountability.

    About Cisco

    Cisco (NASDAQ: CSCO) is the global chief in IT that helps businesses enmesh the alternatives of tomorrow by proving that wonderful issues can pick plot if you connect the up to now unconnected. For ongoing news, tickle fade to http://thenetwork.cisco.com.

    #  #  #

    Cisco and the Cisco emblem are emblems or registered emblems of Cisco and/or its affiliates within the U.S. and different countries. a list of Cisco's emblems can subsist discovered at www.cisco.com/go/logos. Third-celebration emblems mentioned are the property of their respective owners. the exercise of the note companion doesn't imply a partnership relationship between Cisco and another enterprise. 


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    Implementing Cisco IP Telephony and Video, fraction 2 (CIPTV2)

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    Book Extract: Implementing Cisco Unified Communications | killexams.com existent questions and Pass4sure dumps

    Chapter 1, Identifying Issues in a Multisite Deployment

    Deploying Cisco Unified Communications Manager between multiple sites requires an appropriate dial plan, appropriate bandwidth, a character of service (QoS) implementation and a design that can survive IP WAN failures.

    This chapter will define the issues that can arise in a multisite Cisco UC Manager deployment, and recommend specific solutions.

    Identifying Issues in a Multisite DeploymentTable of contents:1. Multisite deployment challenge overview2. character challenges3. Bandwidth challenges4. Availability challenges5. Dial scheme challenges6. Overlapping and nonconsecutive numbers7. Fixed versus variable-length numbering plans8. Variable-length numbering, E.164 addressing and DID9. Optimized call routing and PSTN backup10. NAT and security issues

    Multisite deployment challenge overview

    Formerly known as Cisco call Manager (CCM), Cisco Unified Communications Manager (CUCM) multi-site deployment challenges can include the following:

  • Quality issues: Real-time communications of voice and video must subsist prioritized over a packet-switching network. every sole traffic is treated equally by default in routers and switches. Voice and video are delay-sensitive packets that exigency to subsist given priority to avoid delay and jitter (variable delay), which would result in decreased voice quality.
  • Bandwidth issues: Cisco Unified Communications (Cisco UC) can include voice and video streams, signaling traffic, management traffic, and application traffic such as affluent media conferencing. The additional bandwidth that is required when deploying a Cisco Unified Communications solution has to subsist calculated and provisioned for to ensure that data applications and Cisco Unified Communications applications conclude not overload the available bandwidth. Bandwidth reservations can subsist made to applications through QoS deployment.
  • Availability issues: When deploying Cisco Unified Communications Manager (CUCM) with centralized call processing, IP Phones register with CUCM over the IP LAN and potentially over the WAN. If gateways in remote sites are using Media Gateway Control Protocol (MGCP) as a signaling protocol, they besides depend on the availability of CUCM acting as an MGCP call agent. It is vital to implement fallback solutions for IP Phones and gateways in scenarios in which the connection to the CUCM servers is broken because of IP WAN failure. Fallback solutions besides apply to H.323 gateways but are already created with H.323 dial peers in a proper H.323 gateway configuration.
  • Dial scheme issues: Directory numbers (DN) can overlap across multiple sites. Overlapping dial plans and nonconsecutive numbers can subsist solved by designing a robust multisite dial plan. Avoid overlapping numbers across sites whenever practicable for an easier design.
  • NAT and security issues: The exercise of private IP addresses within an enterprise IP network is very common. Internet Telephony Service Providers (ITSP) require unique public IP addresses to route IP Phone calls. The private IP addresses within the enterprise maintain to subsist translated into public IP addresses. Public IP addresses acquire the IP Phones visible from the Internet and therefore topic to attacks.
  • Quality challenges

    IP networks were not originally designed to carry real-time traffic; instead, they were designed for resiliency and weakness tolerance. Each packet is processed separately in an IP network, sometimes causing different packets in a communications stream to pick different paths to the destination. The different paths in the network may maintain a different amount of packet loss, delay, and delay variation (jitter) because of bandwidth, distance, and congestion differences. The destination must subsist able to receive packets out of order and resequence these packets. This challenge is solved by the exercise of Real-Time Transport Protocol (RTP) sequence numbers and traffic resequencing. When possible, it is best to not depend solely on these RTP mechanisms. Proper network design, using Cisco router Cisco Express Forwarding (CEF) switch cache technology, performs per-destination load sharing by default. Per-destination load sharing is not a perfect load-balancing paradigm, but it ensures that each IP stream (voice call) takes the selfsame path.

    Bandwidth is shared by multiple users and applications, whereas the amount of bandwidth required for an individual IP stream varies significantly during short lapses of time. Most data applications are very bursty, whereas Cisco real-time audio communications with RTP exercise the selfsame continuous-bandwidth stream. The bandwidth available for any application, including CUCM and voice-bearer traffic, is unpredictable. During peak periods, packets exigency to subsist buffered in queues waiting to subsist processed because of network congestion. Queuing is a term that anyone who has ever experienced air flight is close with. When you arrive at the airport, you must acquire in a line (queue), because the number of ticket agents (bandwidth) available to check you in is less than the stream of traffic arriving at the ticket counters (incoming IP traffic). If congestion occurs for too long, the queue (packet buffers) gets filled up, and passengers are annoyed (packets are dropped). Higher queuing delays and packet drops are more likely on highly loaded, slow-speed links such as WAN links used between sites in a multisite environment. character challenges are common on these types of links, and you exigency to wield them by implementing QoS. Without the exercise of QoS, voice packets suffer delay, jitter, and packet loss, impacting voice quality. It is faultfinding to properly configure Cisco QoS mechanisms discontinue to discontinue throughout the network for proper audio and video performance.

    This chapter was excerpted with permissions from Implementing Cisco Unified Communications Manager, fraction 2, by Chris Olsen, Copyright 2008. For more information about this engage and other titles, tickle visit Cisco Press.

    During peak periods, packets cannot subsist sent immediately because of interface congestion. Instead, the packets are temporarily stored in a queue, waiting to subsist processed. The amount of time the packet waits in the queue, called the queuing delay, can vary greatly based on network conditions and traffic arrival rates. If the queue is full, newly received packets cannot subsist buffered anymore and acquire dropped (tail drop). pattern 1-1 illustrates tail drop. Packets are processed on a first in, first out (FIFO) model in the hardware queue of every sole router interfaces. Voice conversations are predictable and constant (sampling is every 20 milliseconds by default), but data applications are bursty and greedy. Voice therefore is topic to degradation of character because of delay, jitter, and packet loss.

    Figure 1-1

    Click for a larger image

    Bandwidth challenges

    Each site in a multisite deployment usually is interconnected by an IP WAN, or occasionally by a metropolitan-area network (MAN) such as Metro Ethernet. Bandwidth on WAN links is limited and relatively expensive. The goal is to exercise the available bandwidth as efficiently as possible. Unnecessary traffic should subsist removed from the IP WAN links through content filtering, firewalls, and access control lists (ACL). IP WAN acceleration methods for bandwidth optimization should subsist considered as well. Any period of congestion could result in service degradation unless QoS is deployed throughout the network.

    Voice streams are constant and predictable for Cisco audio packets. Typically, the G.729 codec is used across the WAN to best exercise bandwidth. As a comparison, the G.711 audio codec requires 64 kbps, whereas packetizing the G.711 voice sample in an IP/UDP/RTP header every 20 ms requires 16 kbps plus the Layer 2 header overhead.

    Voice is sampled every 20 ms, resulting in 50 packets per second (pps). The IP header is 20 bytes, whereas the UDP header is 8 bytes, and the RTP header is 12 bytes. The 40 bytes of header information must subsist converted to bits to pattern out the packet rate of the overhead. Because a byte has 8 bits, 40 bytes * 8 bits in a byte = 320 bits. The 320 bits are sent 50 times per second based on the 20-ms rate (1 millisecond is 1/1000 of a second, and 20/1000 = .02). So:

    .02 * 50 = 1 second320 bits * 50 = 16,000 bits/sec, or 16 kbps

    Voice packets are benign compared to the bandwidth consumed by data applications. Data applications can fill the entire maximum transmission unit (MTU) of an Ethernet frame (1518 bytes or 9216 bytes if jumbo Ethernet frames maintain been enabled). In comparison to data application packets, voice packets are very little (60 bytes for G.729 and 200 bytes for G.711 with the default 20-ms sampling rate).

    In pattern 1-2, a conference bridge has been deployed at the main site. No conference bridge exists at the remote site. If three IP Phones at a remote site unite a conference, their RTP streams are sent across the WAN to the conference bridge. The conference bridge, whether using software or hardware resources, mixes the received audio streams and then sends back three unique unicast audio streams to the IP Phones over the IP WAN. The conference bridge removes the receiver's voice from his or her unique RTP stream so that the user does not suffer echo because of the delay of traversing the WAN link and mixing RTP audio streams in the conference bridge.

    Figure 1-2

    Click for a larger image

    Centralized conference resources cause bandwidth, delay, and capacity challenges in the voice network. Each G.711 RTP stream requires 80 kbps (plus the Layer 2 overhead), resulting in 240 kbps of IP WAN bandwidth consumption by this voice conference. If the conference bridge were not located on the other side of the IP WAN, this traffic would not exigency to traverse the WAN link, resulting in less delay and bandwidth consumption. If the remote site had a CUCM region configuration that resulted in calls with the G.729 codec back to the main site, the software conferencing resources of CUCM would not subsist able to coalesce the audio conversations. Hardware conferencing or hardware transcoder media resources in a voice gateway are required to accommodate G.729 audio conferencing. Local hardware conference resources would remove this need. every sole centrally located media resources (Music On Hold [MOH], annunciator, conference bridges, videoconferencing, and media termination points) suffer similar bandwidth, delay, and resource exhaustion challenges.

    Availability challenges

    When deploying CUCM in multisite environments, centralized CUCM-based services are accessed over the IP WAN. Affected services include the following:

  • Signaling in CUCM multisite deployments with centralized call processing: Remote Cisco IP Phones register with a centralized CUCM server. Remote MGCP gateways are controlled by a centralized CUCM server that acts as an MGCP call agent.
  • Signaling in CUCM multisite deployments with distributed call processing: In such environments, sites are connected via H.323 (non-gatekeeper-controlled, gatekeeper-controlled, or H.225) or Session Initiation Protocol (SIP) trunks.
  • Media exchange: RTP streams between endpoints located at different sites.
  • Other services: These include Cisco IP Phone Extensible Markup Language (XML) services and access to applications such as attendant console, CUCM Assistant, and others.
  • Figure 1-3 shows a Unified Communications network in which the main site is connected to a remote site through a centralized call-processing environment. The main site is besides connected to a remote cluster through an intercluster trunk (ICT) representing a distributed call processing environment. The combination of both centralized and distributed call processing represents a hybrid call-processing model in which little sites exercise the CUCM resources of the main site, but large remote offices maintain their own CUCM cluster. On the bottom left of pattern 1-3 is a SIP trunk, typically over a Metro Ethernet connection to an Internet Telephony Service Provider (ITSP). The profit of the SIP trunk is that the ITSP provides the gateways to the PSTN instead of your providing gateways at the main site.

    Figure 1-3

    Click for a larger image

    An IP WAN outage in pattern 1-3 will cause an outage of call-processing services for the remote site connected in a centralized fashion. The remote cluster will not suffer a call-processing outage, but the remote cluster will not subsist able to dial the main site over the IP WAN during the outage. Mission-critical voice applications (voice mail, interactive voice response [IVR], and so on) located at the main site will subsist unavailable to any of the other sites during the WAN outage.

    If the ITSP is using the selfsame links that allow IP WAN connectivity, every sole calls to and from the public switched telephone network (PSTN) will besides subsist unavailable.

    A deployment enjoy the one shown in pattern 1-3 is considered badly designed because of the exigency of IP WAN and PSTN backup.

    Dial scheme challenges

    In a multisite deployment, with a sole or multiple CUCM clusters, dial scheme design requires the consideration of several issues that conclude not exist in single-site deployments:

  • Overlapping numbers: Users located at different sites can maintain the selfsame directory numbers assigned. Because directory numbers usually are unique only within a site, a multisite deployment requires a solution for overlapping numbers.
  • Nonconsecutive numbers: Contiguous ranges of numbers are vital to summarize call-routing information, analogous to contiguous IP address ranges for route summarization. Such blocks can subsist represented by one or a few entries in a call-routing table, such as route patterns, dial peer destination patterns, and voice translation rules, which retain the routing table short and simple. If each endpoint requires its own entry in the call-routing table, the table gets too big, lots of recollection is required, and lookups pick more time. Therefore, nonconsecutive numbers at any site are not optimal for efficient call routing.
  • Variable-length numbering: Some countries, such as the U.S. and Canada, maintain fixed-length numbering plans for PSTN numbers. Others, such as Mexico and England, maintain variable-length numbering plans. A problem with variable-length numbers is that the complete length of the number dialed can subsist determined only by the CUCM route scheme by waiting for the interdigit timeout. Waiting for the interdigit timeout, known as the T.302 timer, adds to the post-dial delay, which may irk users.
  • Direct inward dialing (DID) ranges and E.164 addressing: When considering integration with the PSTN, internally used directory numbers maintain to subsist related to external PSTN numbers (E.164 addressing). Depending on the numbering scheme (fixed or variable) and services provided by the PSTN, the following solutions are common:
  • Each internal directory number relates to a fixed-length PSTN number: In this case, each internal directory number has its own dedicated PSTN number. The directory number can, but does not maintain to, match the least-significant digits of the PSTN number. In countries with a fixed numbering plan, such as the North American Numbering scheme (NANP), this usually means that the four-digit office codes are used as internal directory numbers. If these are not unique, digits of office codes or administratively assigned site codes might subsist added, resulting in five or more digits being used for internal directory numbers.

    Another solution is to not reuse any digits of the PSTN number but to simply map each internally used directory number to any PSTN number assigned to the company. In this case, the internal and external numbers conclude not maintain anything in common. If the internally used directory number matches the least-significant digits of its corresponding PSTN number, significant digits can subsist set at the gateway or trunk. Also, generic external phone number masks, transformation masks, or prefixes can subsist configured. This is just because every sole internal directory numbers are changed to fully qualified PSTN numbers in the selfsame way. Another example is if the internal directory number is composed of parts of the PSTN number and administratively assigned digits such as site codes plus PSTN station codes, or different ranges, such as PSTN station codes 4100 to 4180 that map to directory numbers 1100 to 1180, or totally independent mappings of internal directory numbers to PSTN numbers. In that case, one or more translation rules maintain to subsist used for incoming calls, and one or more calling party transformation rules, transformation masks, external phone number masks, or prefixes maintain to subsist configured.

  • No DID back in fixed-length numbering plans: To avoid the requirement of one PSTN number per internal directory number when using a fixed-length numbering plan, it is common to disallow DID to an extension. Instead, the PSTN trunk has a sole number, and every sole PSTN calls routed to that number are sent to an attendant, auto-attendant, receptionist, or secretary. From there, the calls are transferred to the appropriate internal extension.
  • Internal directory numbers are fraction of a variable-length number: In countries with variable-length numbering plans, a typically shorter "subscriber" number is assigned to the PSTN trunk, but the PSTN routes every sole calls starting with this number to the trunk. The caller can add digits to identify the extension. There is no fixed number of additional digits or total digits. However, there is a maximum, usually 32 digits, which provides the liberty to select the length of directory numbers. This maximum length can subsist less. For example, in E.164 the maximum number is 15 digits, not including the country code. A caller simply adds the appropriate extension to the company's (short) PSTN number when placing a call to a specific user. If only the short PSTN number without an extension is dialed, the call is routed to an attendant within the company. Residential PSTN numbers are usually longer and conclude not allow additional digits to subsist added; the feature just described is available only on trunks.
  • Type of Number (TON) in ISDN: The calling number (the Automatic Number Identification [ANI]) of calls being received from the PSTN can subsist represented in different ways:

    - As a seven-digit subscriber number- As a ten-digit number, including the area code- In international format with the country code in front of the area code

    To standardize the ANI for every sole calls, the format that is used must subsist known, and the number has to subsist transformed accordingly.

  • Optimized call routing: Having an IP WAN between sites with PSTN access at every sole sites allows PSTN toll bypass by sending calls between sites over the IP WAN instead of using the PSTN. In such scenarios, the PSTN should subsist used as a backup path only in case of WAN failure. Another solution, which extends the idea of toll bypass and can potentially reduce toll charges, is to besides exercise the IP WAN for PSTN calls. With tail-end hop-off (TEHO), the IP WAN is used as much as possible, and the gateway that is closest to the dialed PSTN destination is used for the PSTN breakout.
  • Overlapping and nonconsecutive numbers

    In pattern 1-4, Cisco IP Phones at the main site exercise directory numbers 1001 to 1099, 2000 to 2157, and 2365 to 2999. At the remote site, 1001 to 1099 and 2158 to 2364 are used. These directory numbers maintain two issues. First, 1001 to 1099 overlap; these directory numbers exist at both sites, so they are not unique throughout the complete deployment. This causes a problem: If a user in the remote site dialed only the four digits 1001, which phone would ring? This issue of overlapping dial plans needs to subsist addressed by digit manipulation. In addition, the nonconsecutive exercise of the purview 2000 to 2999 (with some duplicate numbers at the two sites) would require a significant number of additional entries in call-routing tables because the ranges can hardly subsist summarized by one (or a few) entries.

    Figure 1-4

    Click for a larger image

    Fixed versus variable-length numbering plans

    A fixed numbering scheme features fixed-length area codes and local numbers. An open numbering scheme features variance in length of area code or local number, or both, within the country.

    Table 1-1 contrasts the NANP and a variable-length numbering scheme -- Germany's numbering scheme in this example.

    Table 1-1

    Component Description Fixed Numbering scheme (NANP) Variable-Length Numbering scheme (Germany) Country code A code of one to three digits is used to achieve the particular telephone system for each nation or special service. Obtain the E.164 benchmark from http://itu.org to observe every sole international country codes. 1 49 Area code Used within many nations to route calls to a particular city, region, or special service. Depending on the nation or region, it may besides subsist called a numbering scheme area, subscriber trunk dialing code, national destination code, or routing code. Three digits Three to five digits Subscriber number Represents the specific telephone number to subsist dialed, but it does not include the country code, area code (if applicable), international prefix, or trunk prefix. Three-digit exchange code plus a four-digit station code Three or more digits Trunk prefix The initial digits to subsist dialed in a domestic call, before the area code and the subscriber number. 1 0 Access code A number that is traditionally dialed first "to acquire out to the PSTN," used in PBXs and VoIP systems. 9 0 International prefix The code dialed before an international number (country code, area code if any, and then subscriber number). 011 00 or + (+ is used by cell phones)

    Examples:

  • Within the U.S.: 9-1-408-555-1234 or 1-555-1234 (within the selfsame area code)
  • U.S. to Germany: 9-011-49-404-132670
  • Within Germany: 0-0-404-132670 or 0-132670 (within the selfsame area code)
  • Germany to the U.S.: 0-00-1-408-555-1234 (Note: the 1 in 00-1-408 is the U.S. country code, not the trunk prefix.)
  • The NANP PSTN number is 408-555-1234, DID is not used, and every sole calls placed to the main site are handled by an attendant. There is a remote site in Germany with the E.164 PSTN number +49 404 13267. Four-digit extensions are used at the German location, and DID is allowed because digits can subsist added to the PSTN number. When calling the German office attendant (not knowing a specific extension), U.S. users would dial 9-011-49-404-13267. Note how the + is replaced by the international prefix 011 and the access code 9. If the phone with extension 1001 should subsist called directly, 9-011-49-404-13267-1001 has to subsist dialed.

    Variable-length numbering, E.164 addressing and DID

    Figure 1-5 illustrates an example in which the main site with CUCM resides in the U.S. and a remote site without CUCM resides in Germany. The NANP PSTN number in the U.S. is 408-555-1234. Note that DID is not used, because every sole calls placed to the main site are handled by an attendant. A remote site in Germany has PSTN number +49 404 13267. Four-digit extensions are used at the German location, and DID is allowed because digits can subsist added to the PSTN number. When calling the German office attendant (not knowing a specific extension), U.S. users would dial 9-011-49-404-13267. If the phone with extension 1001 should subsist called directly, 9-011-49-404-13267-1001 has to subsist dialed.

    Figure 1-5

    Click for a larger image

    The logic of routing calls by CUCM over the WAN or through the PSTN is appropriately transparent to the phone user.

    Optimized call routing and PSTN backup

    There are two ways to save costs for PSTN calls in a multisite deployment:

  • Total bypass: Calls between sites within an organization that exercise the IP WAN instead of the PSTN. The PSTN is used for intersite calls only if calls over the IP WAN are not possible—either because of a WAN failure or because the call is not admitted by call Admission Control (CAC).
  • Tail-end hop-off (TEHO): Extends the concept of toll bypass by besides using the IP WAN for calls to the remote destinations in the PSTN. With TEHO, the IP WAN is used as much as possible, and PSTN breakout occurs at the gateway that is located closest to the dialed PSTN destination. Local PSTN breakout is used as a backup in case of IP WAN or CAC.
  • In the example shown in pattern 1-6, a call from Chicago to San Jose would subsist routed as follows:

  • The Chicago CUCM Express user dials 9-1-408-555-6666, a PSTN phone located in San Jose.
  • The call is routed from Chicago CUCM Express Router to the San Jose CUCM cluster over the IP WAN with either SIP or H.323.
  • The San Jose CUCM routes the call to the San Jose gateway, which breaks out to the PSTN with what now becomes a local inexpensive call to the San Jose PSTN.
  • The San Jose PSTN Central Office routes the call, and the phone rings.
  • Figure 1-6

    Click for a larger image

    If the WAN were unavailable for any reason before the call, the Chicago Gateway would maintain to subsist properly configured to route the call with the appropriate digit manipulation through the PSTN at a potentially higher toll cost to the San Jose PSTN phone.

    NAT and security issues

    In single-site deployments, CUCM servers and IP Phones usually exercise private IP addresses because there is no exigency to communicate with the outside IP world. NAT is not configured for the phone subnets, and attacks from the outside are impossible. In multisite deployments, however, IP Security (IPsec) virtual private network (VPN) tunnels can subsist used between sites. The VPN tunnels allow only intersite communication; access to the protected internal networks is not practicable from the outside—only from the other site through the tunnel. Therefore, attacks from the outside are blocked at the gateway. To configure IPsec VPNs, the VPN tunnel must subsist configured to terminate on the two gateways in the different sites. Sometimes this is not possible; for instance, the two sites may subsist under different administration, or perhaps security policies conclude not allow the configuration of IPsec VPNs.

    In such a case, or when connecting to a public service such as an ITSP, NAT has to subsist configured for CUCM servers and IP Phones. Cisco calls this Hosted NAT Traversal for Session edge Controllers.

    In pattern 1-7, Company A and Company B both exercise IP network 10.0.0.0/8 internally. To communicate over the Internet, the private addresses are translated into public IP addresses. Company A uses public IP network A, and Company B uses public IP network B. every sole CUCM servers and IP Phones can subsist reached from the Internet and communicate with each other.

    As soon as CUCM servers and IP Phones can subsist reached with public IP addresses, they are topic to attacks from the outside world, introducing potential security issues.

    Figure 1-7

    Click for a larger image

    This download first appeared at searchunifiedcommunications.com

    Cisco Introduces Safety, Security and Communication Solutions for Urban Environments | killexams.com existent questions and Pass4sure dumps

    SAN JOSE, CA and ORLANDO, FL--(Marketwire -09/20/11)- ASIS 2011, Booth # 3525 -- Cisco (NASDAQ: CSCO - News) today announced a series of safety and security innovations powered by a comprehensive Medianet architecture. Together these innovations back the evolution of Cisco's strategy towards a solutions-oriented services-led approach that will enable Cisco customers and large security integrators to build highly scalable network-based security systems that better protect people, property and faultfinding infrastructure.

    At the ASIS show, Cisco today underscored its continued commitment to open standards by demonstrating video back for ONVIF-compliant cameras. The company is besides for the first time extending its Medianet Services Interface software to certified video surveillance partners to automate and streamline the pass in which Cisco networks recognize, configure and monitor the service levels for third party IP cameras.

    As fraction of Cisco's Smart+Connected Communities initiative, Cisco is besides introducing new communications interoperability solutions and enhancements. Real-time communication is faultfinding in times of emergency. Currently in many cases, firefighters, police officers, emergency medical technicians and public safety professionals communicate on different radio frequencies and with different systems, creating barriers to providing a coordinated and efficient response during crises.

    Key Highlights:

    Enhancements to Cisco Video Surveillance Manager 6.3.2

  • Cisco introduced a new tall density Physical Security Storage series platform with the faculty to manage video recording of up to 120 terabytes of storage, and 60 disks in only 4RU of rack space, and up to 240 terabytes, and 120 disks in only 8U of rack space.
  • Video Surveillance Manager besides now supports third party cameras with extended edge storage; this approach effectively extends the value of VSM 6.3.2 to the edge of the network to intelligently manage offload. Working closely with VideoIQ, the leader and pioneer in distributed storage surveillance solutions, Cisco will subsist integrating VideoIQ's iCVR-HD tall Definition line of cameras, including their onboard storage, into Cisco's Video Surveillance Manager.
  • Cisco today demonstrated ONVIF 2.0 streaming video integrated with Cisco Video Surveillance Manager. For customers with third party IP camera deployments, this capability aims to aid protect existing investments and maximizes their selection of deployable cameras.
  • Cisco IP Interoperability and Communications System (IPICS) 4.5

    Cisco IPICS is a complete IP-based dispatch and incidence-response solution enabling rapid, reliable communication using whatever desktop or mobile device is at hand to provide public safety personnel with continuous, real-time communication as well as streaming video.

  • At ASIS, Cisco is demonstrating back for the latest generation of open IP standards interoperability between P25 radio networks and the IP network infrastructure using P25 ISSI gateway capabilities and a universal media services appliance. P25 offers a number of benefits over traditional analog radio systems including improved coverage, better interoperability, and more efficient exercise of the radio spectrum.
  • Extending medianet to the industry

  • Cisco today announced that its Medianet back has been extended to Cisco and other third party cameras. VideoIQ is the first camera vendor to publicly broadcast its intent to embed Medianet Services Interface software into its cameras to aid automate the pass that its cameras interface with Cisco network switches.
  • The Medianet architecture delivers a smarter network, smarter endpoints, cloud services and shared media services. With Medianet Services Interface software, IP cameras can subsist auto-configured upon network connection, and provided with the specific network bandwidth and security they need. Camera connections can subsist diagnosed and even tested with "synthetic" traffic to ensure a trouble-free deployment.
  • Analog to IP-Encoders

  • Cisco besides announced new 4 and 8 port standalone encoders that complement its existing 16-port chassis-based analog to IP video encoders to further back customers in migrating existing deployments to the IP network.
  • Supporting Quotes

  • Bill Stuntz, vice president and generic manager, Physical Security trade Unit, Cisco, said: "Today's announcements portray key steps they are taking as they evolve their safety and security strategy to a more solutions-oriented, services-led approach. They envisage a greater role for their large security integration partners as they focus on projects with their clients that will pick odds of the Medianet architecture. The security trade is converging on IP and open standards, and Cisco is continuing to collaborate with the industry to underpin this evolution."
  • Ed Bednarcik, president of VideoIQ, said: "Optimizing network traffic by intelligently streaming video based on events and content is a key fraction of both their value proposition and Cisco's Medianet. With up to 500GB of onboard storage in each device, VideoIQ's unique solution enables customers to archive video at maximum frame rate, bit rate and resolution without saturating a network -- making the system more scalable, cost effective and easier to deploy. By supporting Medianet Services Interface software on their cameras, they will subsist able to extend these customer benefits even further by automating faultfinding network configurations and helping ensure peak network performance."
  • Warren O'Connor, National Networks manager - Data, New Zealand Police, said: "Implementing Cisco IPICS has provided New Zealand Police with a national radio channel distribution capability. This significantly improved functionality and provided the flexibility to respond quickly and effectively to rapidly changing emergency response scenarios. Any radio channel can now subsist accessed by the Dispatch Console, Operators Console or IP Phone anywhere in the country. This improved capability has been of particular profit for meeting their response to the earthquake in Christchurch as well as the radio communication requirements for the Rugby World Cup."
  • Markus Nepp, project manager for ASFINAG Maut Service GmbH's Systems Operation Department, the Austrian highway and expressways financing company, said: "We previously experienced issues with retransmitting radio channels via PDH, and as a result they switched to voice over IP technology and headset technology. They currently maintain 60 locations equipped with the system, which includes a Cisco router in each tunnel pile connected to the radio interface, and a router installed in the central tunnel control room. Subsequently they introduced IPICS server, and they can now listen into each radio channel from the monitoring center, and besides converse to each channel directly from a Cisco Unified IP phone. They maintain the potential for cell phones or soft clients to participate, and as an added profit they can now connect emergency services into their talk groups. In the future they are aiming to equip another 50 sites with IPICS. For their emergency system, they besides want to change to the CONET Software UC Radio Suite to integrate with their IPICS server."
  • Federico Palleiro, Special Projects manager, Banco de Cordoba, said: "As one of the largest banks in Argentina, security is a key concern. By utilizing Cisco Physical Security Solutions, including Cisco Video Surveillance 2500 IP cameras and Cisco Video Surveillance Stream Manager software, they were able to modernize their technological infrastructure to reduce operating and maintenance costs while converging telephony, video surveillance and data through a network that allows the remote monitoring of 50 branches and the bank's headquarters."
  • Douglas Watters, manager, enterprise infrastructure, Nippon Express (Global Logistics Solutions), said: "In order to present sophisticated logistical solutions to best suit their clients' everyday needs, security is a paramount concern. With their recent comprehensive Cisco security deployment featuring integrated access control and video surveillance monitoring, they were able to centralize and manage security services, lower total cost of operations and comply with industry regulations with self-possession that they are prepared to effectively address any potential threat that could arise, today and in the future."
  • Johann Putter, corporate manager, Information Services, Africa Middle East, at Aurecon in South Africa, said: "With a network of offices across 23 countries, Aurecon required a scalable, comprehensive access control and video surveillance solution that would enable their mobile employees to subsist productive with authorized privileges regardless of their office location. The Cisco Physical Access Control and Video Surveillance solutions they selected allow the management of pile security to subsist done through a single, centralized interface, while ensuring the protection of their intellectual property and their employees' faculty to collaborate effectively. They direct to extend the solution to additional offices worldwide."
  • Supporting Resources:

    Technorati Tags: Cisco, Physical Security, IPICS, VSM, ONVIF, Physical Access, Video Surveillance, Safety & Security, Medianet

    About Cisco SystemsCisco (NASDAQ: CSCO - News) is the worldwide leader in networking that transforms how people connect, communicate and collaborate. Information about Cisco can subsist found at http://www.cisco.com. For ongoing news, tickle fade to http://newsroom.cisco.com.

    Cisco and the Cisco logo are trademarks or registered trademarks of Cisco and/or its affiliates in the U.S. and other countries. A listing of Cisco's trademarks can subsist found at www.cisco.com/go/trademarks. Third-party trademarks mentioned are the property of their respective owners. The exercise of the word partner does not imply a partnership relationship between Cisco and any other company.

    For direct RSS Feeds of every sole Cisco news, tickle visit "News@Cisco" at the following link:

    http://newsroom.cisco.com/dlls/rss.html



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    References :


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    Blogspot : http://killexamz.blogspot.com/2017/05/killexamscom-300-075-braindumps-and.html
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    weSRCH : https://www.wesrch.com/business/prpdfBU1HWO000UEJB
    Dropmark : http://killexams.dropmark.com/367904/11477167
    Issu : https://issuu.com/trutrainers/docs/300-075
    Wordpress : http://wp.me/p7SJ6L-jP
    Scribd : https://www.scribd.com/document/357088034/Pass4sure-300-075-Practice-Tests-with-Real-Questions
    Dropmark-Text : http://killexams.dropmark.com/367904/12038557
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    Box.net : https://app.box.com/s/z4wnrc35347ns1ju6r7ve5j9clbax8mu
    zoho.com : https://docs.zoho.com/file/4b1e134532997e2dc4c44bfa2ce84c2af6ee5






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